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The Evolution of WebRTC

BY UberConference,

So, what is Web Real Time Communication (WebRTC), anyway? The idea isn’t new but people who use voice and video conferencing are beginning to hear it every day. At UberConference we use it to make it easy to join conference calls over the Internet from anywhere.

WebRTC allows real-time voice, video, and data to stream between two people using a web browser. There’s no need for plugins or third-party software, only the latest Chrome or Firefox.

Until recently, web browsers were where you did pretty much everything except conferencing – web surfing, email, watching videos. However, the biggest disadvantage of the web browser was that it was lousy at two-way voice and video calls.

That kind of real-time communication had been challenging for companies for many years because the voice and video compression-decompression algorithms (called codecs) were really expensive. Only a few companies owned them, and they charged pricey licensing fees. In addition, browsers could only request data or send it upon request, not send and receive it in real time, as video chat requires.

To understand this, consider that browsers have been evolving ever since their creation to allow us to do more and more on the web. Plugins were introduced in the mid-late ’90s, which allowed developers to play videos with flash, facilitating a move toward the beginning of video communication. Then, in 2004, the browser language HTML5 developed the <audio> and <video> tags to allow this multimedia content to live in your browser without the need of a plugin. However, real time communication (RTC) remained a challenge because browsers lacked a method to send and receive data in real time, and often the stumbling block were the expensive codecs used to interpret the media communications between users.

For WebRTC to be truly effective, everyone needed access to the high quality codecs. In 2010 Google took on the challenge and purchased two companies: GIPS and On2. This turned the VoIP market on its head.

Here’s why: GIPS was a leading provider of VoIP codecs, On2 had a video codec that rivaled the H.26 standard. And Google open sourced them both, giving the RTC industry a giant push forward.

To solve the media transmission problem, the WebRTC collation created a set of open protocols for browsers to expose to developers. As browsers adopt this standard and implement them, developers can quickly write RTC applications with a few lines of JavaScript code.

That’s why WebRTC has been a big deal for UberConference and for all Internet users. It lets them conference in real-time without having to mess around with applications or phones or leave their web browser.

This is a huge benefit for emerging companies, who, ten years ago, would have paid significantly higher costs for hardware and services to set everything up. They can now build their companies with a much lighter – and cheaper- footprint. Now that’s something to call your CEO about.